[asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup

[Olle E. Johansson]

23 feb 2010 kl. 03.18 skrev Kevin P. Fleming:

> Kirill 'Big K' Katsnelson wrote:
> 
>> The caveat here is that it is perfectly normal NOT to transmit any RTP
>> data in case of long silence. This is why the SIP timers were introduced
>> in the first place: there is no correct way to detect when the client is
>> going away, as no activity is a good session state.
> 
> That's only true when Asterisk tells the other endpoint that it is
> allowed to use voice activity detection and silence suppression, which
> at this point it does not do. In spite of that, there are many endpoints
> that do it anyway, which then causes strange problems on calls,
> including calls getting dropped if an RTP timeout is in use.
Well, the headers we use are note really standardized, at least I could not find them.
In the RTP rfc's it's perfectly legal to just have gaps in the timestamps and stop
sending. However, as both me and Kevin stated, Asterisk does not support it.
On most phones, you can disable silence suppression in the configuration.

/O